search for: option_debug

...c: Device 'Zap/8' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Nov 27 16:46:26] DEBUG[907] chan_zap.c: Exception on 27, channel 8 - this doesn't look good... what does it mean? :-O [Nov 27 16:46:26] DEBUG[907] chan_zap.c: option_debug=100 [Nov 27 16:46:26] DEBUG[907] chan_zap.c: Got event Hook Transition Complete(12) on channel 8 (index 0) [Nov 27 16:46:26] DEBUG[907] chan_zap.c: Sent deferred digit string: Twwww0734414119w [Nov 27 16:46:30] DEBUG[907] chan_zap.c: Exception on 27, channel 8 [Nov 27 16:46:30] DEBUG[907] chan_zap...

Nothing much to be said.. I backported ringinuse, autofill and the QueueLog application from 1.4.0 to 1.2.14. Any or all may be applied - order doesn't matter. They have received minimal testing but appear to function correctly. As always with these things, don't blame me if they connect your callers to a phonesex line, etc. http://bum.net/patches/ Cheers, Gavin.

...| > | | | > | |found = 1; | > | | | > | |if (option_debug > 2) | > | | | > | |ast_log(LOG_DEBUG, "RateMangement: %s\n", s); | > | |...

...| > | | | > | |found = 1; | > | | | > | |if (option_debug > 2) | > | | | > | |ast_log(LOG_DEBUG, "RateMangement: %s\n", s); | > | |...

Hi, Does Asterisk support encryption of voice traffic? I found following wiki that describes IAX RSA authentication. I was able to implement the public/private key authentication among three Asterisk servers connected using IAX protocol. I am not certain if voice traffic can also be encrypted among the Asterisk servers. Your help is highly appreciated.

Hi, I'm trying to get asterisk running on kernel 2.4.20 however trawling through the archives I've found a few references to patches to remove i4l's dtmf detection, but have been unable to find the patch itself (I think it is isdn_audio.c). Can anyone point me in the right direction? The problem I'm seeing is connecting a SIP softphone (tried a few) to an external number via an

...r, totally bail out... */ if (!p->t38.jointcapability || !udptlportno) { ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n"); /* Do NOT Change current setting */ return -1; } else { if (option_debug > 2) ast_log(LOG_DEBUG, "Have T.38 but no audio codecs, accepting offer anyway\n"); } } As I understand if t38 is globally enabled p->t38.jointcapability and udptlportno are always true even so the call is never rejected. As this behavior caused me...

[image: Post]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vdmlld3RvcGljLnBocD9wPTI4NjU%3D&b=2#28652>Posted: Thu Apr 09, 2009 8:34 pm Post subject: DTMF and IVR ... Sorry but URGENT[image: Reply with quote]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vcG9zdGluZy5waHA%2FbW9kZT1xdW90ZSZwPTI4NjUy&b=2>

*Code:* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing Playback("Local/91441425477394 at default-b9f2,1", "sip-silence") in new stack -- Playing 'sip-silence' (language 'en') -- Executing AGI("Local/91441425477394 at default-b9f2,1", "agi:// 127.0.0.1:4577/call_log") in new stack -- AGI Script